The present invention relates generally to a system for allowing devices connected to a network (e.g., an IP or Ethernet network) to collaborate with other such devices so as to transmit and receive data packets without impairment on the network
As is generally known, Ethernet and Internet Protocol (IP) are systems for transmitting packets between different points on a communications network. These switching systems are known as “contention-based” systems. That is, all transmitters contend for network resources. All transmitters may transmit simultaneously. If they do, then network resources may be oversubscribed. When this happens, data may be delayed or lost, resulting in network impairment.
As illustrated in FIG. 1, four streams of packets are input to a packet switch 112, which routes the packets to one or more outputs based on addressing information contained in each packet. Packets may arrive at the switch at unpredictable times, leading to bursts of inputs that must be handled. The switch typically maintains a packet queue 114 that is able to store a small number of packets. The queue may comprise multiple queues arranged by packet priority level, such that priority 3 packets, for example, take precedence over priority 1 packets. If the inputs are too bursty, the queues fill up and some packets may be discarded. The higher-priority queues are typically emptied before the lower-priority queues, such that the lower-priority queues are more likely to lose data first.
IP systems suffer from impairments such as packet loss and jitter. This happens because there is no control over how many such packets reach a router at any given instant. If two packets arrive at a router at the same time, destined for the same port, one will have to be delayed. Both cannot be transmitted simultaneously. One of the packets will be saved in the queue until the first packet is completely transmitted.
FIG. 2 shows a computer network comprising endpoints 100, 101, 102, and 103. The network includes routers 104 through 107. As can be seen in the figure, if endpoints 100 and 101 communicate with endpoints 102 and 103 at the same time, a bottleneck may develop between routers 105 and 106. This may occur because too many packets may be simultaneously transmitted between the routers, causing the routers to discard overflow packets. This can happen even at low levels of network utilization.
Various methods have been developed to overcome data loss on Ethernet and IP networks. The primary approach has been to use additional protocols to replace lost data. This is an after-the-fact solution. An example is the well-known Transmission Control Protocol (TCP). TCP is able to detect data loss and it causes retransmission of the data, until a perfect copy of the complete data file is delivered to the recipient device.
Many devices may be unable to use TCP or any retransmission method because it is far too slow. Real-time applications require delivery of data, accurately, the first time. For these applications to operate well, even the speed of light causes undesired delay. It is not feasible or desirable to add retransmission delay.
The problem is determining how to provide reliable, first-time delivery on a contention-based network. Various approaches have been tried. The most commonly proposed system relies on prioritization of data in the network. With this approach, data having real-time constraints is identified with priority coding so that it may be transmitted before other data.
Prioritization seems at first to be a good solution. However, on reflection it suffers from the same difficulty. Prioritization only provides a delivery advantage relative to the lower-priority data. It provides no advantage against the other priority data. Analysis and testing shows that this approach can work in certain circumstances, but only when the amount of priority data is small. For simple applications like voice, the percentage of the total may need to be 8% or less. Other applications must occupy an even smaller percentage of total network resource. As shown in FIG. 1, even high-priority packets may be discarded if too many high-priority packets are transmitted within a short time interval. For many networks this makes prioritization impractical.
Another approach is to multiplex the data. With this method the bursts of data associated with one flow of data are separated from the burst of another. Multiplexing usually uses some type of time-domain system (known as Time Domain Multiplexing (TDM)) to separate flows. Flows may be separated in groups, so that one group does not contend with another group. This can be an improvement but still leaves the possibility of contention between groups. The only way to eliminate contention is to multiplex each flow individually. A central problem with multiplexing is that it eliminates a principal advantage of the network, namely that average bandwidth available to all is reduced. In other words, each potential transmitter on the network is guaranteed a slot of time on the network, even if that time is infrequently used. This leads to inefficient resource usage.
Asynchronous Transfer Mode (ATM) is another technology for multiplexing a data network, to reduce contention. ATM breaks all data flows into equal length data blocks. Further, ATM can limit the number of data blocks available to any flow or application. The result is a virtual TDM multiplex system.
Both TDM and ATM provide contention reduction, but at the cost of considerable added complexity, cost, components, and lost bandwidth performance. Other approaches rely on specialized hardware to schedule packet delivery, driving up hardware costs.